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DMR Stereo TDMA

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DMR Stereo TDMA
--play voice in both TDMA slots with the -T option (dsd -fr -T)
--Improved MS voice (mostly working) and data handling (hit or miss)
--pretty pretty colors on the console log output
--rate 3/4 data decoding partially working
This commit is contained in:
lwvmobile
2022-04-30 04:43:03 -04:00
committed by GitHub
parent 8b48328431
commit 3cd1eefad8
29 changed files with 3291 additions and 1003 deletions
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551
src/dsd_audio.c
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@@ -21,8 +21,6 @@ pa_sample_spec ss;
pa_sample_spec tt;
pa_sample_spec zz;
pa_sample_spec cc;
//pa_channel_map channel_map;
void openPulseOutput(dsd_opts * opts)
{
@@ -34,68 +32,39 @@ void openPulseOutput(dsd_opts * opts)
tt.format = PA_SAMPLE_S16NE;
tt.channels = opts->pulse_digi_out_channels; //doing tests with 2 channels at 22050 for 44100 audio default in pulse
tt.rate = opts->pulse_digi_rate_out; //48000, switches to 8000 when using RTL dongle
//fprintf (stderr,"digi rate out = %d\n", opts->pulse_digi_rate_out);
//pa_channel_map_init_stereo(&channel_map);
//ss
if (opts->monitor_input_audio == 1)
//if (opts->frame_dmr == 1 && opts->audio_in_type != 3)
{
opts->pulse_raw_dev_out = pa_simple_new(NULL, "DSD-FME", PA_STREAM_PLAYBACK, NULL, "Raw Audio", &ss, NULL, NULL, NULL);
//opts->pulse_raw_dev_out = pa_simple_new(NULL, "DSD FME", PA_STREAM_PLAYBACK, NULL, "DMR/MOTOTRBO Right", &ss, NULL, NULL, NULL);
opts->pulse_raw_dev_out = pa_simple_new(NULL, "DSD-FME", PA_STREAM_PLAYBACK, NULL, "Raw Audio Out", &ss, NULL, NULL, NULL);
}
//tt
pa_channel_map* left = 0; //NULL and 0 are same in this context
pa_channel_map* right = 0; //NULL and 0 are same in this context
//pa_channel_map* right;
//pa_channel_map_init(right);
opts->pulse_digi_dev_out = pa_simple_new(NULL, "DSD-FME", PA_STREAM_PLAYBACK, NULL, opts->output_name, &tt, left, NULL, NULL);
//opts->pulse_digi_dev_out = pa_simple_new(NULL, "DSD FME", PA_STREAM_PLAYBACK, NULL, "Digi Audio Out LEFT", &tt, left, NULL, NULL);
//opts->pulse_raw_dev_out = pa_simple_new(NULL, "DSD-FME", PA_STREAM_PLAYBACK, NULL, "Digi Audio Out RIGHT", &tt, right, NULL, NULL);
if (opts->dmr_stereo == 0)
{
opts->pulse_digi_dev_out = pa_simple_new(NULL, "DSD-FME", PA_STREAM_PLAYBACK, NULL, opts->output_name, &tt, left, NULL, NULL);
}
if (opts->dmr_stereo == 1)
{
opts->pulse_digi_dev_out = pa_simple_new(NULL, "DSD-FME", PA_STREAM_PLAYBACK, NULL, "DMR/MOTOTRBO LEFT", &tt, left, NULL, NULL);
opts->pulse_digi_dev_outR = pa_simple_new(NULL, "DSD-FME", PA_STREAM_PLAYBACK, NULL, "DMR/MOTOTRBO RIGHT", &tt, right, NULL, NULL);
}
}
void openPulseInput(dsd_opts * opts)
{
zz.format = PA_SAMPLE_S16NE;
zz.channels = opts->pulse_raw_in_channels;
zz.rate = opts->pulse_raw_rate_in; //48000
cc.format = PA_SAMPLE_S16NE;
cc.channels = opts->pulse_digi_in_channels;
cc.rate = opts->pulse_digi_rate_in; //48000
//zz
if (opts->monitor_input_audio == 1)
{
//opts->pulse_raw_dev_in = pa_simple_new(NULL, "DSD-FME", PA_STREAM_RECORD, NULL, "Raw Audio In", &zz, NULL, NULL, NULL);
}
//cc
opts->pulse_digi_dev_in = pa_simple_new(NULL, "DSD-FME", PA_STREAM_RECORD, NULL, opts->output_name, &cc, NULL, NULL, NULL);
}
//playRawAudio may be obsoleted now that I can run these correctly right inside of dsd_symbol
void playRawAudio(dsd_opts * opts, dsd_state * state)
{
short obuf, outl, i, sample2, something;
something = state->samplesPerSymbol / 1; //1 for buffer steal loops, 5 for rtl loop
//something = 1;
if (opts->audio_in_type == 0 && opts->audio_out_type == 0){ //hack, but might as well for this particular type since its nearly perfect
for (i=0; i < something; i++){
//read (opts->audio_in_fd, &sample2, 2); //reading here seems to get same speed as gfsk modulation
//pa_simple_drain(opts->pulse_raw_dev_out, NULL);
//pa_simple_read(opts->pulse_raw_dev_in, &state->pulse_raw_out_buffer, 2, NULL );
//obuf = state->pulse_raw_out_buffer / 4; //dividing here 'quietens' it down a little bit, helps with clicking and clipping
//obuf = state->input_sample_buffer;
//obuf = state->pulse_raw_out_buffer;
//write (opts->audio_out_fd, (void*)&obuf, 2);
//pa_simple_write(opts->pulse_raw_dev_out, (void*)&obuf, 2, NULL);
//pa_simple_write(opts->pulse_digi_dev_out, (void*)&obuf, 2, NULL);
//pa_simple_drain(opts->pulse_raw_dev_out, NULL);
}
}
} //end playRawAudio
void
processAudio (dsd_opts * opts, dsd_state * state)
@@ -109,7 +78,6 @@ processAudio (dsd_opts * opts, dsd_state * state)
// detect max level
max = 0;
state->audio_out_temp_buf_p = state->audio_out_temp_buf;
//state->audio_out_temp_buf_pR = state->audio_out_temp_bufR;
for (n = 0; n < 160; n++)
{
aout_abs = fabsf (*state->audio_out_temp_buf_p);
@@ -118,24 +86,18 @@ processAudio (dsd_opts * opts, dsd_state * state)
max = aout_abs;
}
state->audio_out_temp_buf_p++;
//state->audio_out_temp_buf_pR++;
}
*state->aout_max_buf_p = max;
//*state->aout_max_buf_pR = max;
state->aout_max_buf_p++;
//state->aout_max_buf_pR++;
state->aout_max_buf_idx++;
//state->aout_max_buf_idxR++;
if (state->aout_max_buf_idx > 24)
{
state->aout_max_buf_idx = 0;
state->aout_max_buf_p = state->aout_max_buf;
}
//if (state->aout_max_buf_idxR > 24)
// {
// state->aout_max_buf_idxR = 0;
// state->aout_max_buf_pR = state->aout_max_bufR;
// }
// lookup max history
for (i = 0; i < 25; i++)
@@ -241,6 +203,145 @@ processAudio (dsd_opts * opts, dsd_state * state)
state->audio_out_idx2++;
}
}
}
void
processAudioR (dsd_opts * opts, dsd_state * state)
{
int i, n;
float aout_abs, max, gainfactor, gaindelta, maxbuf;
if (opts->audio_gainR == (float) 0)
{
// detect max level
max = 0;
state->audio_out_temp_buf_pR = state->audio_out_temp_bufR;
for (n = 0; n < 160; n++)
{
aout_abs = fabsf (*state->audio_out_temp_buf_pR);
if (aout_abs > max)
{
max = aout_abs;
}
state->audio_out_temp_buf_pR++;
}
*state->aout_max_buf_pR = max;
state->aout_max_buf_pR++;
state->aout_max_buf_idxR++;
if (state->aout_max_buf_idxR > 24)
{
state->aout_max_buf_idxR = 0;
state->aout_max_buf_pR = state->aout_max_bufR;
}
// lookup max history
for (i = 0; i < 25; i++)
{
maxbuf = state->aout_max_bufR[i];
if (maxbuf > max)
{
max = maxbuf;
}
}
// determine optimal gain level
if (max > (float) 0)
{
gainfactor = ((float) 30000 / max);
}
else
{
gainfactor = (float) 50;
}
if (gainfactor < state->aout_gainR)
{
state->aout_gainR = gainfactor;
gaindelta = (float) 0;
}
else
{
if (gainfactor > (float) 50)
{
gainfactor = (float) 50;
}
gaindelta = gainfactor - state->aout_gainR;
if (gaindelta > ((float) 0.05 * state->aout_gainR))
{
gaindelta = ((float) 0.05 * state->aout_gainR);
}
}
gaindelta /= (float) 160;
}
else
{
gaindelta = (float) 0;
}
if(opts->audio_gainR >= 0)
{
// adjust output gain
state->audio_out_temp_buf_pR = state->audio_out_temp_bufR;
for (n = 0; n < 160; n++)
{
*state->audio_out_temp_buf_pR = (state->aout_gainR + ((float) n * gaindelta)) * (*state->audio_out_temp_buf_pR);
state->audio_out_temp_buf_pR++;
}
state->aout_gainR += ((float) 160 * gaindelta);
}
// copy audio data to output buffer and upsample if necessary
state->audio_out_temp_buf_pR = state->audio_out_temp_bufR;
if (opts->split == 0) //shouldn't need splitR
{
for (n = 0; n < 160; n++)
{
upsample (state, *state->audio_out_temp_buf_pR);
state->audio_out_temp_buf_pR++;
state->audio_out_float_buf_pR += 6;
state->audio_out_idxR += 6;
state->audio_out_idx2R += 6;
}
state->audio_out_float_buf_pR -= (960 + opts->playoffsetR);
// copy to output (short) buffer
for (n = 0; n < 960; n++)
{
if (*state->audio_out_float_buf_pR > 32767.0F)
{
*state->audio_out_float_buf_pR = 32767.0F;
}
else if (*state->audio_out_float_buf_pR < -32768.0F)
{
*state->audio_out_float_buf_pR = -32768.0F;
}
*state->audio_out_buf_pR = (short) *state->audio_out_float_buf_pR;
state->audio_out_buf_pR++;
state->audio_out_float_buf_pR++;
}
state->audio_out_float_buf_pR += opts->playoffsetR;
}
else
{
for (n = 0; n < 160; n++)
{
if (*state->audio_out_temp_buf_pR > 32767.0F)
{
*state->audio_out_temp_buf_pR = 32767.0F;
}
else if (*state->audio_out_temp_buf_pR < -32768.0F)
{
*state->audio_out_temp_buf_pR = -32768.0F;
}
*state->audio_out_buf_pR = (short) *state->audio_out_temp_buf_pR;
state->audio_out_buf_pR++;
state->audio_out_temp_buf_pR++;
state->audio_out_idxR++;
state->audio_out_idx2R++;
}
}
}
void
@@ -310,183 +411,77 @@ playSynthesizedVoice (dsd_opts * opts, dsd_state * state)
ssize_t result;
if (state->audio_out_idx > opts->delay)
{
// output synthesized speech to sound card
{
// output synthesized speech to sound card
if(opts->audio_out_type == 2)
{
#ifdef USE_PORTAUDIO
PaError err = paNoError;
do
{
long available = Pa_GetStreamWriteAvailable( opts->audio_out_pa_stream );
if(available < 0)
err = available;
//fprintf (stderr,"Frames available: %d\n", available);
if( err != paNoError )
break;
if(available > SAMPLE_RATE_OUT * PA_LATENCY_OUT)
{
//It looks like this might not be needed for very small latencies. However, it's definitely needed for a bit larger ones.
//When PA_LATENCY_OUT == 0.500 I get output buffer underruns if I don't use this. With PA_LATENCY_OUT <= 0.100 I don't see those happen.
//But with PA_LATENCY_OUT < 0.100 I run the risk of choppy audio and stream errors.
fprintf (stderr,"\nSyncing voice output stream\n");
err = Pa_StopStream( opts->audio_out_pa_stream );
if( err != paNoError )
break;
}
err = Pa_IsStreamActive( opts->audio_out_pa_stream );
if(err == 0)
{
fprintf (stderr,"Start voice output stream\n");
err = Pa_StartStream( opts->audio_out_pa_stream );
}
else if(err == 1)
{
err = paNoError;
}
if( err != paNoError )
break;
err = Pa_WriteStream( opts->audio_out_pa_stream, (state->audio_out_buf_p - state->audio_out_idx), state->audio_out_idx );
if( err != paNoError )
break;
} while(0);
if( err != paNoError )
{
fprintf (stderr, "An error occured while using the portaudio output stream\n" );
fprintf (stderr, "Error number: %d\n", err );
fprintf (stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
}
#endif
//go F yourself PA
}
else
if (opts->monitor_input_audio == 1){
//pa_simple_flush(opts->pulse_raw_dev_in, NULL);
//pa_simple_flush(opts->pulse_raw_dev_out, NULL);
//state->pulse_raw_out_buffer = 0;
//pa_simple_write(opts->pulse_raw_dev_out, (void*)&state->pulse_raw_out_buffer, 2, NULL);
}
//two slot audio testing, still need to seperate channels first internally, but this will play them out of different streams
/*
if(state->currentslot == 0 && opts->audio_in_type != 3)
if(state->currentslot == 0 && opts->audio_in_type != 3 && opts->dmr_stereo == 1)
{
pa_simple_write(opts->pulse_digi_dev_out, (state->audio_out_buf_p - state->audio_out_idx), (state->audio_out_idx * 2), NULL); //Yay! It works.
}
if(state->currentslot == 1 && opts->audio_in_type != 3)
{
pa_simple_write(opts->pulse_raw_dev_out, (state->audio_out_buf_p - state->audio_out_idx), (state->audio_out_idx * 2), NULL); //Yay! It works.
//pa_simple_write(opts->pulse_raw_dev_out, (state->audio_out_buf_pR - state->audio_out_idxR), (state->audio_out_idxR * 2), NULL); //Yay! It works.
state->audio_out_idx = 0;
}
if (opts->audio_in_type == 3)
//if input type is rtl dongle or no dmr stereo...do I need it this way still?
if (opts->audio_in_type == 3 || opts->dmr_stereo == 0)
{
pa_simple_write(opts->pulse_digi_dev_out, (state->audio_out_buf_p - state->audio_out_idx), (state->audio_out_idx * 2), NULL); //Yay! It works.
state->audio_out_idx = 0;
}
*/
pa_simple_write(opts->pulse_digi_dev_out, (state->audio_out_buf_p - state->audio_out_idx), (state->audio_out_idx * 2), NULL); //Yay! It works.
state->audio_out_idx = 0;
}
if (state->audio_out_idx2 >= 800000)
{
state->audio_out_float_buf_p = state->audio_out_float_buf + 100;
state->audio_out_buf_p = state->audio_out_buf + 100;
memset (state->audio_out_float_buf, 0, 100 * sizeof (float));
memset (state->audio_out_buf, 0, 100 * sizeof (short));
state->audio_out_idx2 = 0;
}
{
state->audio_out_float_buf_p = state->audio_out_float_buf + 100;
state->audio_out_buf_p = state->audio_out_buf + 100;
memset (state->audio_out_float_buf, 0, 100 * sizeof (float));
memset (state->audio_out_buf, 0, 100 * sizeof (short));
state->audio_out_idx2 = 0;
}
}
#ifdef USE_PORTAUDIO
int getPADevice(char* dev, int input, PaStream** stream)
void
playSynthesizedVoiceR (dsd_opts * opts, dsd_state * state)
{
int devnum = atoi(dev + 3);
fprintf (stderr,"Using portaudio device %d.\n", devnum);
ssize_t result;
PaError err;
if (state->audio_out_idxR > opts->delay)
{
// output synthesized speech to sound card
if(opts->audio_out_type == 2)
{
//go F yourself PA
}
else
int numDevices = Pa_GetDeviceCount();
if( numDevices < 0 )
{
fprintf (stderr, "ERROR: Pa_GetDeviceCount returned 0x%x\n", numDevices );
err = numDevices;
goto error;
}
if( devnum >= numDevices)
{
fprintf (stderr, "ERROR: Requested device %d is larger than number of devices.\n", devnum );
return(1);
}
//two slot audio testing, still need to seperate channels first internally, but this will play them out of different streams
if(opts->dmr_stereo == 1) //state->currentslot == 1 && opts->audio_in_type != 3 && opts->dmr_stereo == 1
{
pa_simple_write(opts->pulse_digi_dev_outR, (state->audio_out_buf_pR - state->audio_out_idxR), (state->audio_out_idxR * 2), NULL); //Yay! It works.
state->audio_out_idxR = 0;
}
}
const PaDeviceInfo *deviceInfo = Pa_GetDeviceInfo( devnum );
/* print device name */
#ifdef WIN32
{ /* Use wide char on windows, so we can show UTF-8 encoded device names */
wchar_t wideName[MAX_PATH];
MultiByteToWideChar(CP_UTF8, 0, deviceInfo->name, -1, wideName, MAX_PATH-1);
wprintf ( L"Name = %s\n", wideName );
}
#else
fprintf (stderr, "Name = %s\n", deviceInfo->name );
#endif
if((input == 1) && (deviceInfo->maxInputChannels == 0))
{
fprintf (stderr, "ERROR: Requested device %d is not an input device.\n", devnum );
return(1);
}
if((input == 0) && (deviceInfo->maxOutputChannels == 0))
{
fprintf (stderr, "ERROR: Requested device %d is not an output device.\n", devnum );
return(1);
}
//Create stream parameters
PaStreamParameters parameters;
parameters.device = devnum;
parameters.channelCount = 1; /* mono */
parameters.sampleFormat = paInt16; //Shorts
parameters.suggestedLatency = (input == 1) ? PA_LATENCY_IN : PA_LATENCY_OUT;
parameters.hostApiSpecificStreamInfo = NULL;
//Open stream
err = Pa_OpenStream(
stream,
(input == 1) ? &parameters : NULL,
(input == 0) ? &parameters : NULL,
(input == 1) ? SAMPLE_RATE_IN : SAMPLE_RATE_OUT,
PA_FRAMES_PER_BUFFER,
paClipOff,
NULL /*callback*/,
NULL );
if( err != paNoError ) goto error;
return 0;
error:
fprintf (stderr, "An error occured while initializing a portaudio stream\n" );
fprintf (stderr, "Error number: %d\n", err );
fprintf (stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
if (state->audio_out_idx2R >= 800000)
{
state->audio_out_float_buf_pR = state->audio_out_float_bufR + 100;
state->audio_out_buf_pR = state->audio_out_bufR + 100;
memset (state->audio_out_float_bufR, 0, 100 * sizeof (float));
memset (state->audio_out_bufR, 0, 100 * sizeof (short));
state->audio_out_idx2R = 0;
}
}
#endif
void
openAudioOutDevice (dsd_opts * opts, int speed)
{
// get info of device/file
/*
if(strncmp(opts->audio_out_dev, "pulse", 5) == 0)
{
opts->audio_out_type == 0;
}
*/
if(strncmp(opts->audio_out_dev, "pa:", 3) == 0)
{
opts->audio_out_type = 2;
@@ -518,70 +513,6 @@ openAudioOutDevice (dsd_opts * opts, int speed)
fprintf (stderr,"Error, %s is not a device. use -w filename for wav output.\n", opts->audio_out_dev);
exit(1);
}
/*
#ifdef SOLARIS
sample_info_t aset, aget;
opts->audio_out_fd = open (opts->audio_out_dev, O_WRONLY);
if (opts->audio_out_fd == -1)
{
fprintf (stderr,"Error, couldn't open %s\n", opts->audio_out_dev);
//exit (1);
}
// get current
ioctl (opts->audio_out_fd, AUDIO_GETINFO, &aset);
aset.record.sample_rate = speed;
aset.play.sample_rate = speed;
aset.record.channels = 1;
aset.play.channels = 1;
aset.record.precision = 16;
aset.play.precision = 16;
aset.record.encoding = AUDIO_ENCODING_LINEAR;
aset.play.encoding = AUDIO_ENCODING_LINEAR;
if (ioctl (opts->audio_out_fd, AUDIO_SETINFO, &aset) == -1)
{
fprintf (stderr,"Error setting sample device parameters\n");
exit (1);
}
#endif
#if defined(BSD) && !defined(__APPLE__)
int fmt;
opts->audio_out_fd = open (opts->audio_out_dev, O_WRONLY);
if (opts->audio_out_fd == -1)
{
fprintf (stderr,"Error, couldn't open %s\n", opts->audio_out_dev);
opts->audio_out = 0;
//exit(1);
}
fmt = 0;
if (ioctl (opts->audio_out_fd, SNDCTL_DSP_RESET) < 0)
{
fprintf (stderr,"ioctl reset error \n");
}
fmt = speed;
if (ioctl (opts->audio_out_fd, SNDCTL_DSP_SPEED, &fmt) < 0)
{
fprintf (stderr,"ioctl speed error \n");
}
fmt = 0;
if (ioctl (opts->audio_out_fd, SNDCTL_DSP_STEREO, &fmt) < 0)
{
fprintf (stderr,"ioctl stereo error \n");
}
fmt = AFMT_S16_LE;
if (ioctl (opts->audio_out_fd, SNDCTL_DSP_SETFMT, &fmt) < 0)
{
fprintf (stderr,"ioctl setfmt error \n");
}
#endif
*/
}
fprintf (stderr,"Audio Out Device: %s\n", opts->audio_out_dev);
}
@@ -661,90 +592,8 @@ openAudioInDevice (dsd_opts * opts)
{
// this is a device, use old handling, pulse audio now
opts->audio_in_type = 0;
/*
#ifdef SOLARIS
sample_info_t aset, aget;
int rgain;
rgain = 64;
if (opts->split == 1)
{
opts->audio_in_fd = open (opts->audio_in_dev, O_RDONLY);
}
else
{
opts->audio_in_fd = open (opts->audio_in_dev, O_RDWR);
}
if (opts->audio_in_fd == -1)
{
fprintf (stderr,"Error, couldn't open %s\n", opts->audio_in_dev);
//exit(1);
}
// get current
ioctl (opts->audio_in_fd, AUDIO_GETINFO, &aset);
aset.record.sample_rate = SAMPLE_RATE_IN;
aset.play.sample_rate = SAMPLE_RATE_IN;
aset.record.channels = 1;
aset.play.channels = 1;
aset.record.precision = 16;
aset.play.precision = 16;
aset.record.encoding = AUDIO_ENCODING_LINEAR;
aset.play.encoding = AUDIO_ENCODING_LINEAR;
aset.record.port = AUDIO_LINE_IN;
aset.record.gain = rgain;
if (ioctl (opts->audio_in_fd, AUDIO_SETINFO, &aset) == -1)
{
fprintf (stderr,"Error setting sample device parameters\n");
exit (1);
}
#endif
*/
/*
#if defined(BSD) && !defined(__APPLE__)
int fmt;
if (opts->split == 1)
{
opts->audio_in_fd = open (opts->audio_in_dev, O_RDONLY);
}
else
{
opts->audio_in_fd = open (opts->audio_in_dev, O_RDWR);
}
if (opts->audio_in_fd == -1)
{
fprintf (stderr,"Error, couldn't open %s\n", opts->audio_in_dev);
opts->audio_out = 0;
}
fmt = 0;
if (ioctl (opts->audio_in_fd, SNDCTL_DSP_RESET) < 0)
{
fprintf (stderr,"ioctl reset error \n");
}
fmt = SAMPLE_RATE_IN;
if (ioctl (opts->audio_in_fd, SNDCTL_DSP_SPEED, &fmt) < 0)
{
fprintf (stderr,"ioctl speed error \n");
}
fmt = 0;
if (ioctl (opts->audio_in_fd, SNDCTL_DSP_STEREO, &fmt) < 0)
{
fprintf (stderr,"ioctl stereo error \n");
}
fmt = AFMT_S16_LE;
if (ioctl (opts->audio_in_fd, SNDCTL_DSP_SETFMT, &fmt) < 0)
{
fprintf (stderr,"ioctl setfmt error \n");
}
#endif
*/
}
}
}
if (opts->split == 1)